similar to: Cisco 7206 and Sample configs (Newbie)

I see a lot of comments but no actual show runs. Can someone post a 7206 config. I am having a dickens of a time getting calls to pass. I currently have the following loaded. Cisco IOS Software, 7200 Software (C7200-IK9O3S-M), Version 12.3(8)T6, RELEASE SOFTWARE (fc2) Thanks !!! Jerry -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 /

I am running an Asterisk server through a Cisco 7206 PSTN gateway. I am able to make outgoing SIP calls without a problem, though incoming calls have been somewhat of a problem. I am not sure exactly how sip.conf should look in such a scenario. I believe most Cisco gateways are just managed through ACL's, with no authentication, so I think I have the outgoing "peer" statement

All, I know you can use, say, a 2620 w/2 port FXO card as a SIP gateway. Clearly you can use the 5300, 5800, and MGX8850 too. Does anyone know which cards, if any, exist for a 7206VXR to act in a similar capacity, either as a T1/PRI, DS3, or POTS FXO/FXS? What other Cisco routers can act as SIP gateways today? Thanks, Dave

How can ordinary polynomial coefficients be calculated from an orthogonal polynomial fit? I'm trying to do something like find a,b,c,d from lm(billions ~ a+b*decade+c*decade^2+d*decade^3) but that gives: "Error in eval(expr, envir, enclos) : Object "a" not found" > decade <- c(1950, 1960, 1970, 1980, 1990) > billions <- c(3.5, 5, 7.5, 13, 40) > #

Hello.. I have a customer who wants to connect 2 PBX's over IP.. The setup should look like this: [PBX] <-- SS7 --> [Asterisk] <-- IAX --> [Asterisk] <-- SS7 --> [PBX] Since there are no SS7 cards , I was thinking at a way of carrying the E1 data as bulk...Can I do that ? How ? Is possible a scenario like this ? I'm thinking of IAX because I don't

Hello, One of our client company is providing hosted contact center solutions with Cisco IPCC. To keep the Call Recording cost at low, they are planning to use Asterisk / Free Switch. Can anyone integrate Cisco IPCC with Asterisk for call recording ? Regards, Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email:

Hi. I''m having a problem with my testing. The following table in my development database (MySQL 4.1.12, InnoDB): +-----------------------+------------------------+------+-----+---------+----------------+ | Field | Type | Null | Key | Default | Extra | +-----------------------+------------------------+------+-----+---------+----------------+ |

From: Sylvain Gault <sylvain.gault at gmail.com> For the compression / decompression to succeed, the sections layout must be the same between the virtual memory and load memory. The section alignment was kept in sync by introducing aligment that should be greater or equal to the actual section alignment. This patch compute the load memory addresses of the sections so that the layout is the

From: Sylvain Gault <sylvain.gault at gmail.com> TL;DR: The section aligment in linker scripts messed-up the memory mapping needed for the compression / decompression to work. The bug with gcc 5 is not trivial, I'll do my best to explain it here. Basically, there are two memory mappings of the code. One in "virtual memory", and one in "load memory". The one in

N_conreq:entry { self->x=1; calledaddr=(struct xaddrf *)arg3; callingaddr=(struct xaddrf *)arg4; trace(calledaddr->link_id); tracemem(calledaddr->DTE_MAC.lsap_add, 80); trace(callingaddr->link_id); tracemem(callingaddr->DTE_MAC.lsap_add, 80); } 0 -> N_conreq 255

I would like to download climate data files from PCMDI website using R. I tried this line below and I was not able to get the file mainly due to user name and password requirements. I am looking for help for setting up the user and password within R (or somewhere). I have read the FAQ but unfortunately I am a newbie on R and couldnt figure out how to do it. Many thanks in advance

Hello, hope this isn't too far offtopic here but being a troller for a long time here I've realized there is a great knowledge base so I wanted to at least see if i could get some tips. I help run a small colocation company in California and I am in the middle of recommending a new 'core router' platform for our network. We offer mainly colo and dedicated servers, and several of

Hi, My work with Rails has been fairly straightforward, until yesterday... I''ve encountered rather a peculiar problem that I''m not able to solve myself as a rails-newbie. The model code looks like this: class Link < ActiveRecord::Base has_one :user_vote end class UserVote < ActiveRecord::Base belongs_to :user belongs_to :link end I have a SQL query that is

Hi List, I didn''t like the way store_location worked in rails. It doesn''t know you''ve pushed the "back" button, and it always returns to the top of a page. On a large page, the part you were looking at would no longer be on the screen. People try to deal with this using an AJAX hack, but it seems to me that the proper way to fix it is to store the return

Hello folks. I'm very interested in improving LLVM's OCaml bindings. I have several nontrivial patches sitting on llvm-commits for several weeks, and so far there's been little interest in them. Could someone with a good understanding of OCaml please take a look at these? 1) http://llvm-reviews.chandlerc.com/D1925 Every other function in OCaml bindings accepts context

I use cisco ATA 186 ( Version: v2.16 ) with sip protocol and asterisk as pbx. I need feature called as 'three way calling' or 'transfer with consultation'. Registering,calling and 'blind transfer' work fine. Is this feature provided by sip clients or by asterisk itself ? What I have to configure in ATA and what keys I have to press on my phones ? Three way calling is

hi guys, i'm in need of several 24port or higher fxs device which supports sip, aside from mediatrix and audiocodes (cisco's vg248 doesn't support sip), do you have any idea who else manufactures such device? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030801/67eb12dd/attachment.htm

Hi, I'm using rsync version 2.5.5 on OpenBSD 3.3 to backup the users /home directory over the Internet. It's started by a cron job every night. For logging I've add the -vv parameter and redirect the output to a logfile. Like this: rsync -vv --stats --archive --delete --backup --backup-dir=/home/backupdir --compress --rsh 'ssh -2 -i keyfile.key' /home

From: Sylvain Gault <sylvain.gault at gmail.com> Some linker scripts were splitted into i386 and x86_64 versions in commit d8eede3f2a360163235fad222a0190cd7c5bef38 but older scripts were left there. Signed-off-by: Sylvain Gault <sylvain.gault at gmail.com> --- core/syslinux.ld | 414 ------------------------------------------------ core/x86_64/syslinux.ld | 389

'morning everybody, Here is the setup: 5126800422 called 3035 (3035 is a Cisco 7960). The call is g729. 3035 presses 'Conference' on her phone and calls 8327549222. This call is ulaw. (65.72.107.2 is our Cisco 7206 SIP->PRI gateway.) asterisk*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format 65.72.107.2 8327549222 1758081f67e